Add send transports to individual webrtc::Call streams.

webrtc/video/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
@@ skipping 221 matching lines @@
 
   FakeNetworkPipe::Config net_config;
   net_config.queue_delay_ms = 500;
   net_config.loss_percent = 5;
   SyncRtcpObserver audio_observer(net_config);
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
                                     channel,
                                     voe_sync,
                                     &audio_observer);
 
-  Call::Config receiver_config(observer.ReceiveTransport());
+  Call::Config receiver_config;
   receiver_config.voice_engine = voice_engine;
-  CreateCalls(Call::Config(observer.SendTransport()), receiver_config);
+  CreateCalls(Call::Config(), receiver_config);
 
   CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
   EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
 
   AudioPacketReceiver voe_packet_receiver(channel, voe_network);
   audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver);
 
   internal::TransportAdapter transport_adapter(audio_observer.SendTransport());
   transport_adapter.Enable();
   EXPECT_EQ(0,
             voe_network->RegisterExternalTransport(channel, transport_adapter));
 
   observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1);
-  CreateMatchingReceiveConfigs();
+  CreateSendConfig(1, observer.SendTransport());
+  CreateMatchingReceiveConfigs(observer.ReceiveTransport());
 
   send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
   if (fec) {
     send_config_.rtp.fec.red_payload_type = kRedPayloadType;
     send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
     receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
     receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
   }
   receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
   receive_configs_[0].renderer = &observer;
@@ skipping 209 matching lines @@
         : SendTest(kLongTimeoutMs),
           tested_load_(tested_load),
           encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {}
 
     void OnLoadUpdate(Load load) override {
       if (load == tested_load_)
         observation_complete_->Set();
     }
 
     Call::Config GetSenderCallConfig() override {
-      Call::Config config(SendTransport());
+      Call::Config config;
       config.overuse_callback = this;
       return config;
     }
 
     void ModifyConfigs(VideoSendStream::Config* send_config,
                        std::vector<VideoReceiveStream::Config>* receive_configs,
                        VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = &encoder_;
     }
 
@@ skipping 201 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc